| 00:07.17 | *** join/#asterisk Vutral (ss@mirbsd/special/Vutral) |
| 00:50.11 | Micc | the default db name for voicemail messages is asterisk in the code, but in the alembic script it is voicemail |
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| 03:25.26 | acovrig | I get this when trying to call from 329 to 202: http://pastebin.com/M4di47Te this just started, after I installed updates... |
| 03:27.03 | WIMPy | Line 59 tell you that something is wrong in your dilplan, but that looks like something that belongs to #freepbx. |
| 03:27.35 | Penguin | ~freepbx |
| 03:27.35 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 03:28.28 | acovrig | WIMPy, thanks |
| 03:31.25 | acovrig | WIMPy, and as suddenly as the issue started, it ended... |
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| 03:42.10 | *** mode/#asterisk [+o mjordan] by ChanServ |
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| 04:21.48 | glaz | finding a VoIP specialist in Montreal, seem impooooooossible! |
| 04:22.02 | aruntomar | even after changing the settings to my country, i still get the us dialtone indicators https://wiki.asterisk.org/wiki/display/AST/Configuring+Localized+Tone+Indications |
| 04:22.05 | aruntomar | how do i fix it. |
| 04:22.49 | WIMPy | Where? |
| 04:23.30 | WIMPy | What kind of phone? How connected to Asterisk? |
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| 04:27.25 | aruntomar | WIMPy: all sip clients on same lan as the server, using either twinkle or zoiper as the softphone. |
| 04:28.06 | WIMPy | Then you can configure your Asterisk as much as you want. It won't make a difference. It's not involved. |
| 04:28.15 | WIMPy | You need to configutre the phones. |
| 04:28.25 | aruntomar | WIMPy: i had asterisk 1.8.x which was working fine. then i upgraded it to 11 and then to 13. asterisk working fine, only thing is that ringtones are messed up. |
| 04:28.38 | aruntomar | oh |
| 04:29.01 | WIMPy | Ok, with ringtones it can be both. |
| 04:29.08 | aruntomar | WIMPy: but i had not configured any country specific settings in the sip client earlier. |
| 04:29.25 | WIMPy | assumes it's about the ringback tone. |
| 04:29.32 | aruntomar | yes |
| 04:31.03 | aruntomar | WIMPy: i had configured indications.conf, and in chan_dahdi also, i changed the load_zone and default_zone to my country, still it gives the default US ringback tones. |
| 04:31.57 | aruntomar | do i need to compile ringtones/ringback tones for specific countries in asterisk? |
| 04:33.27 | WIMPy | Hmm. If they come form Asterisk, I think it's only about the cahnnels language. |
| 04:34.35 | aruntomar | WIMPy: how do i check if the tones are generated by asterisk or telco provider? |
| 04:35.30 | WIMPy | Or your phone. |
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| 06:55.58 | awk | hi, please assist. http://pastebin.com/Nckpnef6 ... I'm not sure where on the Polycom the digest can be added |
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| 06:56.58 | awk | I'm also getting this |
| 06:56.59 | awk | From: "1202" <sip:172.31.100.20@172.31.100.20>;tag=581D27D0-97940E73 |
| 06:56.59 | awk | To: <sip:172.31.100.20@172.31.100.20> |
| 06:56.59 | awk | CSeq: 2 REGISTER |
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| 07:05.19 | ChannelZ | It's the username |
| 07:05.29 | ChannelZ | What that is in the polycom config, I have no idea |
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| 07:15.40 | awk | found the problem, thanks |
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| 08:36.56 | wasanzy | helllo |
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| 08:49.32 | wasanzy | I still need help |
| 08:56.41 | ChannelZ | We need more to go on. |
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| 09:02.35 | wasanzy | I am using confbridge and I want to be able to allow users in the conference for just few minutes and exit them automatically |
| 09:02.43 | wasanzy | I still can't find anything on that |
| 09:02.50 | wasanzy | I need help please |
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| 09:13.09 | doop | wasanzy why do you want to do that |
| 09:14.21 | ChannelZ | I don't think there's any built-in mechanism to do that. You'd probably have to write an AMI script to watch people entering the conference, time them, and then boot them out. |
| 09:14.50 | doop | Or just dial a local channel and time limit the call |
| 09:15.09 | wasanzy | yes because, I need bill the user every 10 minutes and to do that, I think I have to remove them from the conference after every 10 minutes bill them and put them back to the conference |
| 09:15.28 | ChannelZ | Sex chat? |
| 09:15.40 | wasanzy | doop: yea right, dial, how do I do that? |
| 09:15.56 | ChannelZ | Oh true doop, that might work |
| 09:15.58 | doop | The hell do you mean by bill |
| 09:16.24 | ChannelZ | Take their cash money |
| 09:16.51 | ChannelZ | I MOAN FOR YOU ON PHONE, FIVE DOLLAR! |
| 09:16.58 | doop | Please deposit 25c in your iphoje |
| 09:17.16 | doop | Clonk clonk clonk |
| 09:17.25 | wasanzy | I need to run an external application on the user after every 10 minutes |
| 09:18.39 | wasanzy | How do I use the dial in my scenario? |
| 09:19.34 | ChannelZ | Look up local channels. It's a way to dial the dialplan |
| 09:20.15 | ChannelZ | So instead of calling ConfBridge, you make a utility extension that does it, and then Dial(Local/thingie) |
| 09:20.46 | ChannelZ | (which you can pass appropriate arguments to to limit the call duration) |
| 09:21.01 | wasanzy | like Dial(Local/1,10) ? |
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| 09:23.15 | ChannelZ | Dial(Local/extension@context) |
| 09:23.35 | wasanzy | ok thank you |
| 09:23.57 | ChannelZ | YMMV |
| 09:24.09 | wasanzy | meaning? |
| 09:24.16 | ChannelZ | Your Mileage May Vary |
| 09:25.07 | wasanzy | ok |
| 09:27.52 | *** join/#asterisk Milenco (~Milenco@home.milenco.net) |
| 09:31.52 | Milenco | Just wondering, if I define a context like [anothercontext](+) without defining [anothercontext] first somewhere else..will Asterisk load it of will it fail? |
| 09:45.45 | ChannelZ | do you mean (!) ? Templates? |
| 09:47.58 | Milenco | No..I inherited several Asterisk servers and I want to generalize te config more so its easier for us to maintain |
| 09:48.26 | Milenco | currently we have extensions.conf defining [anothercontext], after that some files get included |
| 09:49.01 | Milenco | these files fill the contexts like [anothercontext](+) followed by the actual settings |
| 09:49.25 | Milenco | I was wondering if I could leave out the [anothercontext] part in extensions.conf |
| 09:49.43 | Milenco | so it would be defined as [anothercontext](+) directly |
| 09:50.16 | Milenco | sorry...contexts=sections |
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| 09:53.00 | ChannelZ | I'd never even heard of the (+). But it adds to an existing context. So presumably the context has to exist to be able to add to it. If you want to get rid of [foo] and only have [foo](+), there's no reason to have [foo](+).. put it all in [foo] |
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| 09:53.27 | ChannelZ | so the opposite of what you're asking, I think. |
| 09:53.58 | Milenco | thats true, but i want to keep it as compatible as possible. meaning i want to define [foo](+) on several places without being afraid they overwrite each other |
| 09:54.58 | ChannelZ | I don't know, try it and find out. Easy enough to do. |
| 09:55.06 | ChannelZ | dialplan show |
| 09:55.43 | Milenco | Thats true..was hoping you guys might know :) But i'll test it |
| 09:55.46 | ChannelZ | my guess is no; the thing you want to add to (+) has to exist in order to add to the thing. |
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| 10:00.18 | Milenco | Too bad, doesn't work: |
| 10:00.18 | Milenco | [Nov 26 10:56:56] WARNING[16532]: config.c:1307 process_text_line: Category addition requested, but category 'incoming' does not exist, line 96 of /etc/asterisk/extensions.d/01_incoming.conf |
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| 10:00.32 | ChannelZ | Occasionally I make good guesses |
| 10:00.54 | Milenco | Thanks for your help ChannelZ :) |
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| 10:11.35 | wasanzy | I added this: Dial(local/join_conf@voicemenu-menu_level0,5) to call that extension which will join the conference and timeout after 5 secs, but it is not timing out, |
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| 10:20.50 | ChannelZ | That's a timeout on ringing. |
| 10:20.56 | ChannelZ | core show application dial |
| 10:21.11 | ChannelZ | One of the arguments is a call timer |
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| 10:41.18 | wasanzy | S(n): Hangup the call n seconds AFTER called party picks up. |
| 10:42.02 | wasanzy | ,Dial(local/join_conf@voicemenu-menu_level0,5,S(5)) will that work?am yet to try though |
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| 10:44.57 | gavimobile | how can I discover the status of a dahdi/zaptel channel if its pluged in or not.. I've tried cat /proc/zaptel/1 && zap show channels && zap show channel X && lszaptel |
| 10:44.57 | gavimobile | <PROTECTED> |
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| 13:45.03 | Ibrahim22 | hi, how do i disable loading acl.conf ? |
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| 13:46.01 | WIMPy | Delete it. |
| 13:46.20 | Ibrahim22 | yeah, but if i do that, it shows WARNING failed to load acl.conf |
| 13:46.54 | [TK]D-Fender | Then ignore it |
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| 15:08.31 | wasanzy | how do I put caller on hold in a conference after n minutes? |
| 15:11.52 | [TK]D-Fender | wasanzy: go make some application that polls *'s active channels, sees which one is on hold, then does an AMI Transfer once the time has been reached |
| 15:18.02 | wasanzy | g option in dial() is to continue executing the next priority in the dialpan under the same or current context after the user's call hangup right? if that i so, I have this: Dial(local/join_conf@voicemenu-menu_level0,5,g,S(10)) |
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| 15:18.36 | wasanzy | but it is hanging up the call after 10 secs and the next priority is not being executed |
| 15:19.27 | [TK]D-Fender | wasanzy: And you aren't using your parameters right |
| 15:19.40 | [TK]D-Fender | Dial does not have 4 paramters |
| 15:19.51 | [TK]D-Fender | "core show application dial" <---- |
| 15:20.43 | wasanzy | so how do I do that? I read the doc already but couldn't place my hand on how to do that |
| 15:20.49 | wasanzy | could u show me? |
| 15:21.56 | [TK]D-Fender | wasanzy: Read the instructions |
| 15:22.14 | [TK]D-Fender | wasanzy: Step 1: Learn how to count the number of commas |
| 15:24.37 | wasanzy | Dial(local/join_conf@voicemenu-menu_level0,5,gS(10)) |
| 15:24.47 | wasanzy | is that right? am testing it though |
| 15:25.38 | Skintkingle | Hi Guys. getting a strange one where an extension is reporting DND but isn't set on the phone. Anyone know how I can find out why the extension is expected to be DND. |
| 15:25.46 | Skintkingle | I get this back from putty: Got SIP response 480 "Do Not Disturb" back from <source ip> |
| 15:27.53 | [TK]D-Fender | It is set to DND |
| 15:28.17 | [TK]D-Fender | Either globally, or as a response to specific contact(s) |
| 15:29.36 | Skintkingle | is there any way i can see set DNDs from within putty? |
| 15:30.13 | [TK]D-Fender | Your phone supports being managed via SSH? |
| 15:31.09 | Skintkingle | oh right... yeah. lol. Having a derp day. That's the response from the phone to asterisk. Ignore that question. lol |
| 15:31.58 | file | responds with 200 Need Lunch |
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| 15:38.57 | *** mode/#asterisk [+o newtonr] by ChanServ |
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| 15:59.25 | vassilux | pickup wroks well if I have a user in the queue , but I use queue just for park an incall and notify users by phone's BLF. Any idea How I can take a call from the queue ? |
| 16:00.12 | [TK]D-Fender | Pickup works on a ringing channel. |
| 16:00.27 | [TK]D-Fender | Which could be a member being called, etc. |
| 16:00.47 | [TK]D-Fender | But there is no "pickup" for a channel sitting in a queue awaiting distribution |
| 16:00.52 | vassilux | Yes , but ... The customer don't want make rings phones |
| 16:01.05 | [TK]D-Fender | All you can do is hijack it via AMI Transfer/Bridge, etc |
| 16:01.42 | [TK]D-Fender | Queue is not an appropriate tool for this, so expect your use of it to require some level of hackery |
| 16:01.59 | vassilux | Yes , I did a module with python but for the actual case it is too short |
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| 16:18.17 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 16:34.27 | Skintkingle | Anyone have any idea why this would be happening when a call is attempting to be made? : WARNING[2373]: chan_sip.c:1980 retrans_pkt: Maximum retries exceeded on transmission 3c2671ca80c6-r9kjsw94g5fe for seqno 1 (Critical Response) |
| 16:35.52 | [TK]D-Fender | You're sending out comms and not getting an answer |
| 16:36.34 | [TK]D-Fender | Due to any number of ciscumstances such as filtering, network handling inconsistencies, system misconfiguration, provider blockage, etc |
| 16:36.39 | *** join/#asterisk riess82 (~riessma@193-81-125-11.adsl.highway.telekom.at) |
| 16:40.16 | *** join/#asterisk gp5st1 (~gp5st@static-96-235-41-234.pitbpa.fios.verizon.net) |
| 16:44.31 | gp5st1 | hello. I'm starting a small call center (~2 seats now and hopefully 6-10 by the end of the year). I'm debating if running my own asterisk server for call queueing or using an external service for that all. Is asterisk difficult to keep running, up-to-date, configured if one hasn't used it before, but is used t keeping other services up and running? |
| 16:44.38 | *** join/#asterisk s7r (~s7r@openvpn/user/s7r) |
| 16:44.52 | *** join/#asterisk linetrace (~linetrace@65.19.81.126) |
| 16:45.10 | [TK]D-Fender | Once * is running, it tends to keep running |
| 16:45.21 | [TK]D-Fender | updating isn't a big deal. |
| 16:45.35 | [TK]D-Fender | As for configuring, that's up to you. |
| 16:45.53 | [TK]D-Fender | * is a PBX and telephony toolkit. It is whatever you make of it. |
| 16:46.19 | [TK]D-Fender | Go download it, install it, and start reading the book to understand it. |
| 16:46.23 | [TK]D-Fender | ~book |
| 16:46.23 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 16:46.38 | linetrace | we just completed an upgrade from asterisk 1.4.x (I know!) to the 12.7.0 and discovered that the 'eventwhencalled' option is no longer available in queues.conf |
| 16:46.44 | gp5st1 | [TK]D-Fender: the toolkit part is what scares me:) I'd be super excited to have more control and the fun of doing it; I'm just concerned I won't be able to make it as reliable as I'd like it |
| 16:47.11 | [TK]D-Fender | gp5st1: Untold thousds use it in production all the time. |
| 16:47.11 | gp5st1 | thanks for the book link:) |
| 16:47.17 | linetrace | we have some cdr reporting software that seems to be missing that data, is there a related config that I've missed? |
| 16:47.19 | Skintkingle | it is a remote phone. so i guess that. |
| 16:47.27 | gp5st1 | [TK]D-Fender: I'm more worried about me, not the quality of * :) |
| 16:47.37 | gp5st1 | I've heard amazing things about * |
| 16:49.53 | [TK]D-Fender | linetrace: that event is AMI, and has nothing to do with CDR |
| 16:50.32 | linetrace | [TK]D-Fender: oh, okay, that wasn't clear in the documentation i was finding, thanks |
| 16:51.08 | linetrace | [TK]D-Fender: Oops, except my reporting software is actually using AMI, not CDR, my bad |
| 16:51.08 | [TK]D-Fender | Queue has nothing to do with CDR's as it is. |
| 16:51.43 | linetrace | so I guess my original question still stands s/cdr/AMI/ |
| 16:56.49 | mdhas | Does anyone know if there's a way to only register sip devices in a particular subnet? say 10.0.0.0/24 and ignore all registeration requests from any other ip. |
| 16:58.26 | [TK]D-Fender | linetrace: Have you checked to see if there is a new PERMANENT event that gets issued? |
| 16:58.47 | [TK]D-Fender | mdhas: permit/deny |
| 16:59.04 | [TK]D-Fender | mdhas: It's all in the sample config. Go read it. |
| 16:59.35 | mdhas | D-Fender: thanks :) |
| 17:02.53 | *** join/#asterisk igcewieling (~ewieling@ip98-170-196-157.pn.at.cox.net) |
| 17:05.46 | *** join/#asterisk sgriepentrog (~sgriepent@nat/digium/x-hxnaljdmubliewee) |
| 17:11.37 | igcewieling | We have AELSub() to call AEL from extensions.conf and return to extensions.conf when done. How would I call extensions.conf "macros" (really gosub) from AEL. IfI try using Gosub() in the AEL the parser compalins. |
| 17:14.19 | *** join/#asterisk anonymouz666 (~anonymouz@189-25-35-215.user.veloxzone.com.br) |
| 17:14.54 | [TK]D-Fender | igcewieling: I suspect you might be able to call them directly with the AEL syntax |
| 17:15.31 | [TK]D-Fender | igcewieling: It should get parsed back as long as the name matches.... |
| 17:16.17 | igcewieling | "[Nov 26 12:15:57] WARNING[27463]: ael/pval.c:2532 check_pval_item: Warning: file /etc/asterisk/extensions.ael, line 205-205: application call to Gosub affects flow of control, and needs to be re-written using AEL if, while, goto, etc. keywords instead! |
| 17:16.17 | igcewieling | " |
| 17:16.51 | [TK]D-Fender | PB the related bits up... |
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| 17:20.40 | igcewieling | If I have to write an extensions.conf version (to call from Dial) and an AEL version (to call from AEL) I can, but it seems silly. |
| 17:21.01 | igcewieling | [TK]D-Fender: here is the actual line, I can try to put together a pastebin, but it is a lot of code |
| 17:21.04 | igcewieling | <PROTECTED> |
| 17:21.36 | igcewieling | I was thinking maybe use stackpush / stackpop and goto to work around the issue. |
| 17:23.23 | [TK]D-Fender | igcewieling: I was saying to use AEL syntax to call the Gosub |
| 17:23.35 | [TK]D-Fender | &contex() or whatever that formatting was |
| 17:34.13 | igcewieling | [Nov 26 12:31:37] ERROR[27834][C-00000b64]: app_stack.c:567 gosub_exec: Attempt to reach a non-existent destination for Gosub: (Context:sm_intercept, Extension:~~s~~, Priority:1) |
| 17:34.46 | igcewieling | daffy-01*CLI> dialplan show sm_intercept |
| 17:34.46 | igcewieling | [ Context 'sm_intercept' created by 'pbx_config' ] |
| 17:34.46 | igcewieling | <PROTECTED> |
| 17:35.42 | [TK]D-Fender | Can you just show us the 2 chunks of dialplan.. something does not feel clear here.... |
| 17:36.14 | [TK]D-Fender | Your request seems to be wanting to call a NON-AEL Gosub target from within AEL. Is this not the case? |
| 17:38.00 | gp5st1 | is it common to run asterisk on freebsd or is linux more common and recommended? |
| 17:39.37 | igcewieling | [TK]D-Fender: you are correct. "How would I call extensions.conf "macros" (really gosub) from AEL. " It looks like I'll have to have two versions, one for extensions.conf and one for extensions.ael |
| 17:40.28 | [TK]D-Fender | igcewieling: Can you show some more complete code for sanity's sake... |
| 17:40.41 | [TK]D-Fender | igcewieling: To make sure we're not missing anything |
| 17:42.01 | igcewieling | [TK]D-Fender: I'm trying. the pastebin I usually use is not working and I need one which expires posts. |
| 17:45.11 | igcewieling | try this http://pastebin.com/q5ttB52h |
| 17:45.14 | igcewieling | [TK]D-Fender: |
| 17:46.26 | [TK]D-Fender | Extension:~~s~~, |
| 17:46.33 | [TK]D-Fender | Well the error line seemed to include a ~ |
| 17:46.45 | [TK]D-Fender | should be able to shove that in your pattern match |
| 17:46.46 | igcewieling | ~~s~~ is done by AEL |
| 17:46.46 | infobot | igcewieling: okay |
| 17:47.07 | igcewieling | I can try it though |
| 17:47.13 | [TK]D-Fender | You'll have to make your dialplan compatible with the AEL call |
| 17:47.31 | [TK]D-Fender | But I would think that it'd be doable |
| 17:50.31 | *** join/#asterisk ph8 (~ph8@unaffiliated/ph8) |
| 17:50.46 | igcewieling | using a sub context in extensions.conf which just calls an AELSub seems like the easiest |
| 17:51.00 | igcewieling | I'd hoped there was an easy answer. 8-( |
| 17:51.07 | [TK]D-Fender | Whatever floats your boat I guess... |
| 17:52.31 | igcewieling | AELsub exists because it is very tough to go between AEL and extensions.conf. I'm not going to fight it. I have more important things to do. |
| 17:52.57 | igcewieling | But thanks for the help, I appreciate it,. |
| 17:54.24 | [TK]D-Fender | np |
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| 18:50.59 | *** join/#asterisk MarkS- (~mark@unaffiliated/mark21) |
| 18:53.04 | MarkS- | Hello, is it possible within voicemail.conf (or in a dialplan before entering the voicemail) to set certain values on a per mailbox or per context base? They are normally located in [general]. eg serveremail, fromstring, emailsubject, emaildateformat, emailbody |
| 18:58.08 | newtonr | I don't believe so |
| 18:58.17 | newtonr | but.. I'm not 100% positive |
| 18:59.31 | newtonr | actually yeah it looks like maybe you can, for some of the options |
| 18:59.45 | newtonr | https://issues.asterisk.org/jira/browse/ASTERISK-13487 |
| 19:00.33 | newtonr | well and in voicemail.conf it says |
| 19:00.35 | newtonr | "; NOTE: All options can be expressed globally in the general section, and |
| 19:00.35 | newtonr | ; overridden in the per-mailbox settings, unless listed otherwise. |
| 19:00.35 | newtonr | " |
| 19:00.52 | newtonr | so yeah! |
| 19:01.56 | *** join/#asterisk Chainsaw (~chainsaw@gentoo/developer/chainsaw) |
| 19:05.55 | MarkS- | newtonr: let me test a few options, however I probably need to test another way as it didn't work in the first tested method |
| 19:09.20 | newtonr | if you can't do what the documentation claims, then make sure to file an issue on the tracker so we can either fix the documentation or fix the bug. |
| 19:14.07 | Milenco | hey hello! I got a macro i use for outbound dialing, but it always exits non-zero |
| 19:14.23 | Milenco | it works just fine but i was wondering how to prevent it |
| 19:14.31 | Milenco | do i need to handle hangups or something? |
| 19:15.38 | MarkS- | newtonr: I can set the following on a per mailbox base: serveremail, emailsubject and emailbody |
| 19:16.18 | MarkS- | The fromstring doesn't work, but let me first upgrade to the latest 13.x version (coming from 11.x) before filing a bug about it |
| 19:25.42 | [TK]D-Fender | Milenco: It exits that way when the call ends |
| 19:25.51 | *** join/#asterisk areski (~areski@80.174.128.55.dyn.user.ono.com) |
| 19:26.16 | Milenco | yes, it seems to happen with all my calls |
| 19:26.40 | Milenco | but a non-zero exit code usually means not-good |
| 19:27.01 | Milenco | and even tho everything works i like to handle it so i won't exit non-zero |
| 19:27.58 | Milenco | http://www.voip-info.org/wiki/view/Asterisk+cmd+Macro mentions that you should add a 'h' extension which i did |
| 19:28.25 | Milenco | which also gets executed, but still returns the non-zero from the original Dial() command |
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| 19:47.06 | *** join/#asterisk sparetire_ (~sparetire@unaffiliated/sparetire) |
| 19:54.28 | [TK]D-Fender | Milenco: Show us your dialplan and a call |
| 19:54.31 | [TK]D-Fender | ~pb |
| 19:54.31 | infobot | A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com, http://pastebin.ca, http://channels.debian.net/paste, http://paste.lisp.org, http://bin.cakephp.org/; or install pastebinit with yum or aptitude. |
| 19:54.32 | [TK]D-Fender | ^^^ |
| 19:56.36 | MarkS- | setting the fromstring on a per mailbox for voicemail would be nice, it seems that that isn't supported at this moment |
| 20:12.17 | *** join/#asterisk ChannelZ (channelz@burner.com) |
| 20:13.16 | MarkS- | https://issues.asterisk.org/jira/browse/ASTERISK-24562 for the created issue, should I add additional information? |
| 20:18.09 | *** join/#asterisk tuc0 (~tuc0@d192-24-85-250.nap.wideopenwest.com) |
| 20:19.08 | tuc0 | Hey guys, proxying calls through a different asterisk server which makes a call out a switch, but I'm getting different SIP response codes passed back than what the switch tells the asterisk proxy |
| 20:20.15 | tuc0 | for instance I'll get a SIP 480 Temporarily Unavailable (Call limit) to the proxy but the proxy will pass back to my local asterisk box SIP response 503 "Service Unavailable" |
| 20:20.36 | tuc0 | Is there something I can do to make it more transparent? |
| 20:24.07 | *** join/#asterisk tzafrir (~tzafrir@bzq-218-28-58.cablep.bezeqint.net) |
| 20:27.38 | dan_j | Hi. I've been asked to put the parameter ;privacy=full into the Remote-Party-ID sip header but can't seem to find any documentation on how to do that. Has anyone tried this? |
| 20:27.39 | [TK]D-Fender | tuc0: You are not proxying your calls through another Asterisk server |
| 20:27.52 | [TK]D-Fender | tuc0: Asterisk is NOT a proxy |
| 20:28.12 | *** part/#asterisk rrittgarn (~rrittgarn@75-150-221-196-Illinois.hfc.comcastbusiness.net) |
| 20:28.34 | [TK]D-Fender | tuc0: You should not expect it to act like one. It is very limited in the kinds of things it can report back, and that's up to you and your dialplan to call the apps to try to set that status to the call |
| 20:29.01 | [TK]D-Fender | dan_j: "core show function CALLERID" <- |
| 20:31.41 | dan_j | [TK]D-Fender: Thanks. Where can I find a definition of what all the different datatypes mean? |
| 20:33.25 | tuc0 | [TK]D-Fender, thank you, can you suggest an app I can use to set that kind of information? |
| 20:34.44 | [TK]D-Fender | dan_j: https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CALLERID |
| 20:35.01 | [TK]D-Fender | tuc0: "core show application hangup" |
| 20:35.32 | dan_j | [TK]D-Fender: I've seen that but it doesnt explain what all the datatypes mean. One example of many... subaddr-odd ? |
| 20:35.34 | tuc0 | [TK]D-Fender, thanks again, I didn't realize hangup could work that way. |
| 20:35.46 | dan_j | Which one is required to set to what for a withheld call? |
| 20:37.08 | [TK]D-Fender | dan_j: should be name-pres and num-pres |
| 20:37.42 | dan_j | Set to what? ""? |
| 20:38.06 | [TK]D-Fender | http://www.voip-info.org/wiki/view/Asterisk+func+callerid |
| 20:39.22 | tuc0 | [TK]D-Fender, is it typical to see Hangup(${HANGUPCAUSE}) then? |
| 20:40.59 | dan_j | Thanks |
| 20:42.25 | tuc0 | I want to make sure I'm not replacing a value in hangup that will confuse or break something down the line if I use it that way. And a bit curious as to why it's not defaulted to the hangupcause |
| 20:44.04 | igcewieling | tuc0: HANGUPCAUSE is set by Dial, not by Hangup. |
| 20:44.21 | igcewieling | Hangup(${HANGUPCAUSE}) seems to be entirely useless. |
| 20:44.41 | tuc0 | not useless, it worked. |
| 20:44.49 | igcewieling | tuc0:worked to do what? |
| 20:44.56 | tuc0 | my not "asterisk proxy" is now passing the correct sip response code |
| 20:45.12 | igcewieling | tuc0: what happens if you try just Hangup() ? |
| 20:45.29 | tuc0 | it was passing a different response code |
| 20:45.47 | igcewieling | Interesting. Another example of Asterisk not being a proxy |
| 20:46.01 | tuc0 | it was giving me a 503 when the proxy got a 480 |
| 20:49.58 | tuc0 | igcewieling, scratch that, it was the missing hangup command itself, not the ${HANGUPCAUSE}. I forgot I added hangup entirely to the end of the context. |
| 20:50.08 | tuc0 | so you were right |
| 20:50.15 | igcewieling | there you gp |
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| 20:50.49 | tuc0 | thanks guys |
| 20:51.49 | igcewieling | using hangup handlers is much more reliable than using the 'h' extension. |
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| 23:23.53 | wasanzy | hi |
| 23:24.05 | ChannelZ | low |
| 23:24.16 | wasanzy | my head is basting |
| 23:24.23 | wasanzy | over this conference issue |
| 23:30.28 | *** join/#asterisk king1337-2 (~king1337@S0106c8fb2641d848.vs.shawcable.net) |
| 23:45.26 | ChannelZ | sounds dirty |
| 23:51.42 | ChannelZ | Dial(Local/conferenceexten@context,,gL(600000)) |
| 23:51.55 | ChannelZ | Playback(give-me-more-moneh) |
| 23:54.05 | wasanzy | ChannelZ: sorry are you still around? |
| 23:54.39 | ChannelZ | yes |
| 23:54.59 | wasanzy | ok |
| 23:55.07 | wasanzy | let me paste my config |